CN102316406B - Audio signal processing apparatus and audio signal processing method - Google Patents
Audio signal processing apparatus and audio signal processing method Download PDFInfo
- Publication number
- CN102316406B CN102316406B CN201110151059.7A CN201110151059A CN102316406B CN 102316406 B CN102316406 B CN 102316406B CN 201110151059 A CN201110151059 A CN 201110151059A CN 102316406 B CN102316406 B CN 102316406B
- Authority
- CN
- China
- Prior art keywords
- loud speaker
- coefficient
- audio signal
- unit
- filter
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R5/00—Stereophonic arrangements
- H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
- H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
Abstract
An audio signal processing apparatus includes a signal processing unit, an output unit, a retention unit, and a coefficient setting unit. The signal processing unit is configured to perform signal processing on an audio signal by a digital filter. The output unit is configured to be connected to an external speaker and output the audio signal to the speaker. The retention unit is configured to retain a plurality of filter coefficients that are impulse responses having reverse characteristics of a plurality of speakers having different speaker characteristics. The coefficient setting unit is configured to select one of the filter coefficients that corresponds to the speaker connected to the output unit from the retention unit and set the filter coefficient in the digital filter.
Description
Technical field
The disclosure relates to audio signal execution correction process to correct audio signal processor and the acoustic signal processing method of loudspeaker performance.
Background technology
In the equipment (hereinafter referred to as audio signal processing apparatus) of the execution Audio Signal Processing of such as stereo set and so on, there is the technology wherein audio signal obtained from sound source being performed to the correction process of such as digital filter process and so on.Audio signal processing apparatus exports the audio signal having stood correction process from loud speaker etc., thus can improve sound quality, acoustics etc. that the audio frequency from loud speaker etc. exports.
The example of this correction process comprises the correction of " loudspeaker performance ".Loudspeaker performance refers to the frequency characteristic of loud speaker, and this frequency characteristic is different from bore (bore) of loud speaker or its internal structure etc.Here, frequency characteristic refers to as the phase characteristic of time deviation, the amplitude characteristic etc. as strength ratio between the audio signal being input to loud speaker and the phase place of audio signal exported from loud speaker.
The example that can correct the audio signal processing apparatus of loudspeaker performance by performing correction process to audio signal comprise such as Japanese Patent Application Publication No.2009-55079 (the 34th section, Fig. 1; Hereinafter referred to as patent documentation 1) disclosed in " signal processing apparatus ".This signal processing apparatus is intended to the low level component that the magnification ratio by the low band signal in conjunction with input audio signal and the frequency displacement to high frequency band thereof improve compact loud speaker.
Summary of the invention
But as in signal processing apparatus disclosed in patent documentation 1, the correction process improving pre-set frequency band is only applicable to the situation of wherein specifying speaker types and the loudspeaker performance that will connect.The example of audio signal processing apparatus comprises and not to be formed with integrated with loudspeaker and any loud speaker is connected to its equipment by user.In this case, even if when audio signal stands with the conventional stereo correction process that speaker types is irrelevant, the effect that will obtain also can be limited or causes reverse effect.
Special in recent years, portable music reclaim equiment etc. is used widely, and user has increasing chance by this equipment connection to optional loud speaker.Such as, widely used base type loud speaker (docking speaker) etc., utilizing this base type loud speaker to carry can from the portable music reclaim equiment of earphone output audio with thus from loud speaker output audio.In this case, the loudspeaker performance change of the loud speaker of audio signal processor will be connected to.
In view of said circumstances, expect to provide a kind of audio signal processor and the acoustic signal processing method that can perform the correction process corresponding with the loudspeaker performance of the loud speaker that will be connected to audio signal.
According to embodiment of the present disclosure, provide a kind of audio signal processor comprising signal processing unit, output unit, holding unit and coefficient setup unit.
Signal processing unit is configured to pass digital filter to the process of audio signal executive signal.
Output unit is configured to be connected to outside loud speaker and audio signal is outputted to loud speaker.
Holding unit is configured to keep multiple filter coefficient, and the plurality of filter coefficient is the impulse response of the opposite characteristic of multiple loud speakers with different loudspeaker performance.
What coefficient setup unit was configured to select the filter coefficient corresponding with the loud speaker being connected to output unit from holding unit sets filter coefficient in the lump digital filter.
According to embodiment of the present disclosure, in holding unit, keep filter coefficient in advance, this filter coefficient is the impulse response of the opposite characteristic of multiple loud speakers with different loudspeaker performance.The impulse response of loud speaker can be measured by providing impulse signal to loud speaker and collecting output audio by microphone, and the opposite characteristic of loud speaker can obtain from the impulse response measured.The impulse response with opposite characteristic is set as filter coefficient makes opposite characteristic give audio signal, and therefore can corrects the loudspeaker performance of the loud speaker corresponding with this filter coefficient.When loud speaker is connected to output unit, coefficient setup unit selects the filter coefficient corresponding with this loud speaker.Coefficient setup unit sets this filter coefficient in the digital filter of signal processing unit.Correspondingly, in the digital filter of signal processing unit, audio signal stands the signal transacting corresponding with the loud speaker being connected to output unit and outputs to this loud speaker from output unit.As mentioned above, audio signal processor can perform the correction process corresponding with the loudspeaker performance of the loud speaker being connected to output unit to audio signal.
Holding unit can also keep the coefficient length of each of the filter coefficient corresponding with the reproduction band of multiple loud speaker, and coefficient setup unit can set filter coefficient with reference to coefficient length in digital filter.
Loud speaker has the lowest resonant frequency determined based on its structure, and loud speaker is difficult to the audio frequency that suitable output frequency is equal to or less than lowest resonant frequency.Therefore, by the correction process of digital filter, be not suitable for correcting the frequency equaling or schedule lowest resonant frequency.Here, the frequency band by determining to correct as the coefficient length of filter coefficient number.In other words, by being that there is the coefficient length corresponding with the reproduction band of loud speaker by filter coefficient setting, only correction process can be performed to the reproduction band of loud speaker.In addition, because the coefficient length being used for correcting the frequency band of the lowest resonant frequency being equal to or less than loud speaker is unnecessary, so the amount of calculation of signal processing unit also can be reduced.
Holding unit can also keep sound channel set information, whether this sound channel set information is corresponding from each loud speaker of multiple loud speaker and indicate filter coefficient to be different between sound channel, and coefficient setup unit can set filter coefficient with reference to sound channel set information in digital filter.
The situation that some of them loud speaker is stereo (two sound channel) with the different L channel of loudspeaker performance and R channel can be envisioned.According to this embodiment of the present disclosure, even if when the loudspeaker performance of sound channel is different, also the correction process corresponding with each sound channel can be performed to audio signal.In addition, when the L channel of loud speaker is identical with the loudspeaker performance of R channel, a filter coefficient can be used in for the correction process of respective speaker and the capacity of holding unit can be saved.
Holding unit can also keep number of channels information, and number of channels information is corresponding with each loud speaker of multiple loud speaker and indicate number of channels, and coefficient setup unit can set filter coefficient with reference to number of channels information in digital filter.
According to embodiment of the present disclosure, according to the number of channels of loud speaker, the correction process for correcting loudspeaker performance is performed to audio signal.Be in monaural situation at loud speaker, the number of channels for digital filter process can be adjusted and reduce amount of calculation.In addition, compared with being stereosonic situation with loud speaker, being filter coefficient can be reduced half in monaural situation at loud speaker, and saving the capacity of holding unit.
Holding unit can also keep loud speaker identification information, loud speaker identification information is corresponding with each loud speaker of multiple loud speaker and be associated with each model of multiple loud speaker, and coefficient setup unit can set the filter coefficient of the loud speaker of the loud speaker identification information be assigned with corresponding to out of Memory in digital filter, this out of Memory obtains from the loud speaker being connected to output unit and indicates the model of loud speaker.
When loud speaker is connected to output unit, in order to coefficient setup unit can select the filter coefficient corresponding with this loud speaker, coefficient setup unit needs the model identifying loud speaker.The input being used to specify loud speaker model that loud speaker model can be made by such as user identifies.But as in embodiment of the present disclosure, coefficient setup unit obtains the information of instruction model from loud speaker and this information is compared with loud speaker type information, and result is that coefficient setup unit can identify loud speaker model when user only connects loud speaker.
The coefficient word length of all right retention coefficient setup unit of holding unit, this coefficient word length is corresponding with each loud speaker of multiple loud speaker, and coefficient setup unit can set filter coefficient by referential digit length in digital filter.
According to embodiment of the present disclosure, according to the coefficient word length of signal processing unit, can the correction process for correcting loudspeaker performance be performed to audio signal and reduce the amount of calculation of signal processing unit.
Audio signal processor can also comprise: test signal output unit, and the loud speaker be configured to being connected to output unit exports test signal; Audio collection unit, is configured to pass test signal and collects the audio frequency exported from loud speaker; And coefficient generation unit, the audio frequency being configured to collect from audio collection unit generates the filter coefficient corresponding with loud speaker, and keeps filter coefficient in holding unit.
According to embodiment of the present disclosure, even if when the filter coefficient loud speaker do not remained in holding unit of its correspondence is connected to output unit, audio signal processor also can generate the filter coefficient corresponding with this loud speaker and use this filter coefficient in correction process.Correspondingly, loudspeaker performance can be corrected for various loud speaker (loud speaker just do not remained in advance in holding unit) according to the audio signal processor of embodiment of the present disclosure.
Audio signal processor can also comprise: test signal output unit, and the loud speaker be configured to being connected to output unit exports test signal; Audio collection unit, is configured to pass test signal and collects the audio frequency exported from loud speaker; And coefficient generation unit, the audio frequency being configured to collect from audio collection unit generates the filter coefficient corresponding with loud speaker, and is associated with the filter coefficient with highest similarity in the filter coefficient kept in holding unit by loud speaker.
According to embodiment of the present disclosure, even if when the filter coefficient loud speaker do not remained in holding unit of its correspondence is connected to output unit, audio signal processor also can generate the filter coefficient corresponding with this loud speaker and use this filter coefficient in correction process.In this case, loud speaker compared with the filter coefficient kept in holding unit, and to be associated with the filter coefficient with highest similarity by coefficient generation unit by newly-generated filter coefficient.It should be noted that similarity can judge based on whether the value of such as filter coefficient is close to each other.Correspondingly, even if also do not add new filter coefficient to holding unit when new loud speaker connects, and the capacity of holding unit can be saved.
According to another embodiment of the present disclosure, provide a kind of acoustic signal processing method, the method comprises the impulse response measured and have multiple loud speakers of different loudspeaker performance.
The filter coefficient obtained from impulse response remains on holding unit, is associated with multiple loud speaker simultaneously.
Select one of filter coefficient corresponding with the loud speaker connected to be set in digital filter from holding unit, and be applied to audio signal.
As mentioned above, according to embodiment of the present disclosure, audio signal processor and the acoustic signal processing method that can perform the correction process corresponding with the loudspeaker performance of the loud speaker connected to audio signal can be provided.
According to the detailed description of following embodiment as shown in the drawings, these and other objects of the present disclosure, Characteristics and advantages will become more obvious.
Accompanying drawing explanation
Fig. 1 is the block diagram of the audio signal processor illustrated according to first embodiment of the present disclosure;
Fig. 2 is the schematic diagram of the example of the digital filter that signal processing unit is shown;
Fig. 3 illustrates the impulse response of particular speaker and the curve chart of frequency characteristic thereof;
Fig. 4 illustrates the impulse response of the opposite characteristic with loud speaker and the curve chart of frequency characteristic thereof;
Fig. 5 illustrates the impulse response of loud speaker and the curve chart of frequency characteristic thereof that obtain after performing correction process to audio signal;
Fig. 6 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of the first embodiment;
Fig. 7 is the example of the menu screen shown over the display by coefficient setup unit;
Fig. 8 is the flow chart of the operation of the audio signal processor illustrated according to the first embodiment;
Fig. 9 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of second embodiment of the present disclosure;
Figure 10 is the impulse response of the loud speaker illustrated for comparing and the curve chart of frequency characteristic thereof;
Figure 11 illustrates the impulse response of the opposite characteristic with loud speaker and the curve chart of frequency characteristic thereof;
Figure 12 illustrates the impulse response of loud speaker and the curve chart of frequency characteristic thereof that obtain after performing correction process to audio signal;
Figure 13 is the flow chart of the operation of the audio signal processor illustrated according to the second embodiment;
Figure 14 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of third embodiment of the present disclosure;
Figure 15 is the flow chart of the operation of the audio signal processor illustrated according to the 3rd embodiment;
Figure 16 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of fourth embodiment of the present disclosure;
Figure 17 is the flow chart of the operation of the audio signal processor illustrated according to the 4th embodiment;
Figure 18 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of fifth embodiment of the present disclosure;
Figure 19 is the flow chart of the operation of the audio signal processor illustrated according to the 5th embodiment;
Figure 20 is the schematic diagram of the coefficient files that the various loud speakers kept are shown in the holding unit according to the audio signal processor of sixth embodiment of the present disclosure;
Figure 21 is the flow chart of the operation of the audio signal processor illustrated according to the 6th embodiment;
Figure 22 is the block diagram of the audio signal processor illustrated according to seventh embodiment of the present disclosure;
Figure 23 is the perspective view of the outward appearance of the audio signal processor illustrated according to the 7th embodiment;
Figure 24 is the perspective view of the audio signal processor according to the 7th embodiment, and the state of wherein being collected audio frequency by microphone is shown;
Figure 25 is the perspective view of the audio signal processor according to the 7th embodiment, and the state of wherein being collected audio frequency by microphone is shown;
Figure 26 is the flow chart of the operation of the audio signal processor illustrated according to the 7th embodiment; And
Figure 27 is the flow chart of the operation of the audio signal processor illustrated according to eighth embodiment of the present disclosure.
Embodiment
(the first embodiment)
First embodiment of the present disclosure will be described.
[structure of audio signal processor]
Fig. 1 is the block diagram of the audio signal processor 1 illustrated according to first embodiment of the present disclosure.Audio signal processor 1 shown in Fig. 1 is such as portable music reclaim equiment.
As shown in Figure 1, audio signal processor 1 comprises acquiring unit 2, signal processing unit 3, output unit 4, holding unit 5 and coefficient setup unit 6.Acquiring unit 2 and output unit 4 are connected to each other via signal processing unit 3, and holding unit 5 is connected to signal processing unit 3 via coefficient setup unit 6.In addition, Fig. 1 shows the loud speaker S and sound source M that are connected with output unit 4.In addition, also loud speaker S can be replaced by frames connecting with headphone.
Acquiring unit 2 obtains audio signal from sound source M.Sound source M can be the sound source that is recorded on the recording medium of such as CD (compact disk) and so on or can be the sound source obtained from internet etc.Acquiring unit 2 can be such as CD driver.The audio signal of acquisition is supplied to signal processing unit 3 by acquiring unit 2.The audio signal obtained by acquiring unit 2 can be analog signal or digital signal.When analog signal, analog signal stands A/D (analog/digital) conversion in acquiring unit 2.
Signal processing unit 3 performs correction process to the audio signal provided from acquiring unit 2.Signal processing unit 3 can be digital filter.The groups of filter coefficients that signal processing unit 3 utilizes the coefficient files of loud speaker S to comprise performs above-mentioned correction process, and this coefficient files is set by coefficient setup unit 6, and its details will be described later.The audio signal standing correction process is supplied to output unit 4 by signal processing unit 3.
The audio signal provided from signal processing unit 3 is exported to loud speaker S by output unit 4.Output unit 4 comprises such as D/A (digital-to-analog) transducer or amplifier.In addition, output unit 4 is provided with the connector that loud speaker S can be attached thereto.Such as, the shape of this connector can limit the model of the loud speaker that can be connected to output unit 4.
Holding unit 5 keeps " coefficient files " of all kinds loud speaker.Holding unit 5 is ROM (read-only memory), RAM (random access memory) etc.
The coefficient files of all kinds loud speaker candidate that coefficient setup unit 6 keeps from holding unit 5 selects the coefficient files of the loud speaker S be connected with output unit 4, and in signal processing unit 3, set the groups of filter coefficients that this coefficient files comprises.In the present embodiment, the information of the coefficient setup unit 6 loud speaker S that uses input unit (not shown) to input based on user selects corresponding coefficient files.
Structure audio signal processor 1 described above.It should be noted that, be not limited to those shown in specification according to the audio signal processor of disclosure embodiment, and comprise the scheme be equal to audio signal processor 1.Such as, more above-mentioned structures can be arranged as the multiple devices be connected to each other.
[digital filter]
The digital filter of signal processing unit 3 will be described now.
Fig. 2 is the schematic diagram of the example of the digital filter that signal processing unit 3 is shown.Fig. 2 shows FIR (finite impulse response) filter, but can use other digital filter of such as IIR (infinite impulse response) filter and so on.
As shown in Figure 2, digital filter F comprises multiple (N number of) delay block 11, multiplier 12 and adder 13.Be input to the input signal Si g of digital filter F
xin delay block 11, stand Z-transformation (Laplace transform about discrete signal) and postpone a clock.Delayed signal in multiplier 12 with predetermined filter coefficient group h (filter coefficient h
0to h
nset) be multiplied.Groups of filter coefficients h determines in the measurement operation that will be described later.Signal through multiplier 12 is exported as output signal Sig by adder 13 phase adduction
y.
The set of the adder 13 that the output of the multiplier 12 that the output of delay block 11, delay block 11 is input to and multiplier 12 is input to is taps (tap) 14.In other words, digital filter F comprises N number of tap 14.Number (hereinafter referred to as number of taps) along with tap 14 becomes large, and frequency characteristic can change more quickly, but the amount of calculation of digital filter F increases.By number (hereinafter referred to as number of taps) and the groups of filter coefficients h of tap 14, determine the filter characteristic of digital filter F.As mentioned above, signal processing unit 3 is applied and is wherein used audio signal as input signal Si g
xdigital filter F, and export calibrated audio signal as output signal Sig
y.
[correction process]
Now the correction of signal processing unit 3 pairs of audio signals will be described through.
As mentioned above, the groups of filter coefficients that the coefficient files that signal processing unit 3 is used in loud speaker S comprises performs by the correction process of digital filter F to audio signal.For this process, pre-determine the groups of filter coefficients h of loud speaker S.
Groups of filter coefficients h determines based on the measurement result of " impulse response " of loud speaker S.The measurement of impulse response uses loud speaker S and performs with preset distance and the right microphone of loud speaker S-phase.Impulse signal (transient audio signal) is supplied to loud speaker S and from loud speaker S output audio.Audio frequency uses microphone to measure to obtain impulse response.Fig. 3 A illustrates the example of measured impulse response.In the curve chart shown in Fig. 3 A, trunnion axis instruction time, vertical axis indicator range.Impulse response shown in Fig. 3 A stands Fourier transform (time-domain signal converts frequency-region signal to), thus obtains the frequency characteristic shown in Fig. 3 B.In the curve chart shown in Fig. 3 B, trunnion axis instruction frequency, vertical axis indicator range.The characteristic of the loud speaker shown in Fig. 3 A and Fig. 3 B is loudspeaker performance.
By the correction process performed by signal processing unit 3, the loudspeaker performance of the loud speaker S shown in Fig. 3 A and Fig. 3 B is corrected to ideal loudspeaker characteristic.Suppose ideal loudspeaker and microphone with the identical distance of distance when measuring the impulse response of loud speaker S toward each other, then ideal loudspeaker characteristic refers to the impulse response and frequency characteristic thereof collected by microphone.Here, as ideal loudspeaker characteristic, the sharp-pointed and loudspeaker performance of frequency characteristic flat in the peak illustrating wherein impulse, but loudspeaker performance is not limited to this, but any loudspeaker performance can be set.
In order to the loudspeaker performance of loud speaker S is corrected to ideal loudspeaker characteristic, only need the filter coefficient h obtaining groups of filter coefficients h
0to h
n, and by digital filter F by this filter coefficient h
0to h
napplied audio signal.For this reason, calculate " opposite characteristic " by using the loudspeaker performance being measured as the loud speaker S of " 1 " to carry out dividing.Fig. 4 A illustrates the impulse response with opposite characteristic, and Fig. 4 B illustrates the frequency characteristic with opposite characteristic.The impulse response with opposite characteristic can be set to the filter coefficient h of digital filter
0to h
n.Filter coefficient h
0to h
nnumber (number of taps) be the peak number order of impulse response.
Signal processing unit 3 performs correction process by the digital filter F wherein setting groups of filter coefficients h as mentioned above to audio signal.Correspondingly, opposite characteristic is endowed audio signal and is superimposed on when by loudspeaker performance during loud speaker S output audio.In other words, the loudspeaker performance of loud speaker S is corrected.Fig. 5 A illustrates the impulse response of the loud speaker S when audio signal stands correction process, and Fig. 5 B illustrates its frequency characteristic.As fig. 5 a and fig. 5b, the peak of impulse response is sharpened and frequency characteristic flattens smooth.
[coefficient files]
As mentioned above, the loudspeaker performance of loud speaker S can use the groups of filter coefficients h obtained from the opposite characteristic of loud speaker S to correct.Therefore, by storing the groups of filter coefficients h of loud speaker S to keep groups of filter coefficients h in holding unit 5 in " coefficient files " that associate with loud speaker S-phase, audio signal processor 1 can correct the loudspeaker performance of the loud speaker S when loud speaker S is connected to output unit 4.
In addition, be similar to loud speaker S, audio signal processor 1 can keep the coefficient files of the groups of filter coefficients h comprising other model loud speaker that can be connected to output unit 4 in holding unit 5.Fig. 6 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.In figure 6, the loud speaker S that model is different is expressed as loud speaker S
a, loud speaker S
bwith loud speaker S
c, and loud speaker S
agroups of filter coefficients h, loud speaker S
bgroups of filter coefficients h and loud speaker S
cgroups of filter coefficients h be expressed as groups of filter coefficients h
a, groups of filter coefficients h
bwith groups of filter coefficients h
c.
[selection of coefficient files]
As mentioned above, select the coefficient files of the loud speaker corresponding with the loud speaker model being connected to output unit 4 in the coefficient files of the various loud speakers that coefficient setup unit 6 keeps in holding unit 5, and in signal processing unit 3, set the groups of filter coefficients h be included in selected coefficient files.Specifically, coefficient setup unit 6 can show choice menus and user be made one's options on the display being supplied to audio signal processor 1.Fig. 7 illustrates the example that will be presented at the menu screen on display D by coefficient setup unit 6.When user inputs the model of the loud speaker connected, coefficient setup unit 6 selects the coefficient files of corresponding loud speaker model.
[operation of audio signal processor]
Now by the operation of description audio signal processing apparatus.
Fig. 8 is the flow chart of the operation that audio signal processor 1 is shown.
As shown in Figure 8, when loud speaker S is connected to output unit 4, coefficient setup unit 6 shows above-mentioned menu screen (St101) over the display.When receiving the operation input made by user, coefficient setup unit 6 selects the coefficient files (St102) of corresponding loud speaker.Next, coefficient setup unit 6 sets the groups of filter coefficients h (St103) be included in this coefficient files in the digital filter F of signal processing unit 3.In this way, audio signal processor 1 sets filter coefficient according to the model of connected loud speaker in the digital filter of signal processing unit 3.
When sending the instruction to reproducing audio, acquiring unit 2 obtains audio signal from sound source M and audio signal is supplied to signal processing unit 3.Signal processing unit 3 performs correction process, so that obtained audio signal is supplied to output unit 4 by using digital filter F to provided audio signal.The process that output unit 4 performs such as D/A conversion or amplifies provided audio signal and so on, and obtained audio signal is supplied to loud speaker S with output audio.When user's change is connected to the loud speaker S of output unit 4, audio signal processor 1 sets the groups of filter coefficients h be included in the coefficient files corresponding with loud speaker model again in digital filter F.
As mentioned above, in the present embodiment, because audio signal processor 1 keeps the coefficient files of all kinds loud speaker that can be attached thereto, so can according to the model setting digital filter of connected loud speaker.Correspondingly, audio signal processor 1 can perform correction process correct loudspeaker performance according to the model of the loud speaker that will connect to audio signal.
(the second embodiment)
Second embodiment of the present disclosure will be described now.
In a second embodiment, by the same reference numeral sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the difference of the present embodiment and the first embodiment is the details of the coefficient files kept in holding unit 5.
[coefficient files]
Fig. 9 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.As shown in Figure 9, corresponding with each loud speaker coefficient files also comprises " filter coefficient length " m except groups of filter coefficients h.Filter coefficient length m is length (the filter coefficient h of groups of filter coefficients h
0to h
nnumber), and for loud speaker S each model and set.In fig .9, loud speaker S
afilter coefficient length m be expressed as filter coefficient length m
a, loud speaker S
bfilter coefficient length m be expressed as filter coefficient length m
b, loud speaker S
cfilter coefficient length m be expressed as filter coefficient length m
c.
The correcting range of filter coefficient length m on loudspeaker performance has impact.As mentioned above, audio signal stands correction process by signal processing unit 3 and the loudspeaker performance of loud speaker S is corrected.But loud speaker has the lowest resonant frequency f0 derived from its schematic diagram, and loud speaker is difficult to the suitable output frequency audio frequency lower than lowest resonant frequency f0.
Figure 10 A is the curve chart of the impulse response of the loud speaker T illustrated for comparing, and Figure 10 B is the curve chart that its frequency response is shown.Figure 11 A is the curve chart of the impulse response that the opposite characteristic with loud speaker T is shown, and Figure 11 B is the curve chart that its frequency response is shown.Figure 12 A illustrates wherein to the curve chart of the impulse response of loud speaker T when audio signal execution correction process, and Figure 12 B is the curve chart that its frequency response is shown.Make loud speaker T and loud speaker S experience same treatment, in other words, measure the impulse response of loud speaker T and loud speaker S and calculate its groups of filter coefficients, then correcting loudspeaker performance by digital filter.
Comparison diagram 3B and Figure 10 B, in the state before loudspeaker performance corrects, can the frequency band of output audio in loud speaker T than wider to arrive lower frequency side in loud speaker S, this reveals that the frequency f 0 of loud speaker T is less than the frequency f 0 of loud speaker S.As shown in Fig. 4 B and Figure 11 B, the frequency band of opposite characteristic is distinguished not quite in low-frequency band.But, as shown in Fig. 5 B and Figure 12 B, loudspeaker performance correct after state in, loudspeaker performance all flattens smooth in both figures, but loud speaker T have more broadband to arrive lower frequency side.
As shown in these figures, because loud speaker has the lowest resonant frequency f0 depending on its structure, so be difficult to be compensated by the correction process of audio signal lower than the frequency band of frequency f 0.In addition, when the audio signal of the frequency band lower than frequency f 0 is supplied to loud speaker, there is a kind of like this worry, namely audio signal does not export as audio frequency, and occurs the nonlinear distortion of such as harmonic distortion and so on.Therefore, just in the frequency band being equal to or greater than frequency f 0, correcting audio signals is applicable to according to loud speaker model.
Here, in digital filter, according to the frequency band of audio signal standing correction process, be included in filter coefficient length m and the filter coefficient h of the necessity in groups of filter coefficients h
0to h
nnumber different.The filter coefficient length needed for audio signal corrected in low-frequency band is greater than the filter coefficient length needed for audio signal corrected in high frequency band.Therefore, can by changing filter coefficient length m to limit according to loud speaker model (lowest resonant frequency f0) by the frequency band standing the audio signal of correction process.In the above example, there is by making the filter coefficient length m of the loud speaker S with high-frequency f0 be less than the filter coefficient length m of the loud speaker S of low frequency f0, correction process can be performed for the frequency band corresponding with each loud speaker to audio signal.
Therefore, give the filter coefficient length m corresponding with the model of this loud speaker by the coefficient files of the loud speaker kept in holding unit 5, coefficient setup unit 6 can from filter coefficient h
0to h
nmiddle selection proper filter coefficients is to be set in the digital filter F of signal processing unit 3.
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 13 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 13, when loud speaker S is connected to output unit 4, coefficient setup unit 6 shows above-mentioned menu screen (St201) over the display.When receiving the operation input made by user, coefficient setup unit 6 selects the coefficient files (St202) of corresponding loud speaker.Next, coefficient setup unit 6 is with reference to the filter coefficient length m (St203) be included in the coefficient files of selected loud speaker.Subsequently, coefficient setup unit 6 sets the proper filter coefficients h in groups of filter coefficients h in digital filter F based on filter coefficient length m
0to h
n(St204).When sending the instruction of reproducing audio, audio signal processor performs correction process with from loud speaker S output audio to audio signal as when the first embodiment in signal processing unit 3.
As mentioned above, in the present embodiment, because coefficient files comprises the filter coefficient length m corresponding with the model of loud speaker S, so only the audio signal of appropriate frequency bands stands correction process in signal processing unit 3.Correspondingly, the audio frequency that frequency can be prevented to equal or equal lowest resonant frequency f0 exports from loud speaker S.In addition, based on filter coefficient length m from filter coefficient h
0to h
nselect proper filter coefficients, and the number of taps of digital filter F reduces.Therefore, the amount of calculation of signal processing unit 3 can also be reduced.
(the 3rd embodiment)
Third embodiment of the present disclosure will be described now.
In the third embodiment, by the identical reference symbol sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the difference of the present embodiment and the first embodiment is the details of the coefficient files kept in holding unit 5.
[coefficient files]
Figure 14 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.As shown in figure 14, corresponding with each loud speaker coefficient files comprises groups of filter coefficients h and " channel information " C.Here, in the situation that the R channel (Rch) of loud speaker is different with the loudspeaker performance of L channel (Lch) wherein, coefficient files comprises the groups of filter coefficients h corresponding with corresponding sound channel.In addition, in the situation that L channel is identical with the loudspeaker performance of R channel wherein, coefficient files comprises the groups of filter coefficients h shared by two sound channels.Here, loud speaker S
bl channel different with the loudspeaker performance of R channel, and loud speaker S
awith loud speaker S
cin the L channel of each identical with the loudspeaker performance of R channel.Channel information C is identical or different information about the groups of filter coefficients used in the L channel of loud speaker and R channel.In fig. 14, loud speaker S
achannel information be expressed as channel information C
a, by loud speaker S
al channel and R channel share groups of filter coefficients be expressed as groups of filter coefficients h
a, this is equally applicable to loud speaker S
c.In addition, loud speaker S
bchannel information be expressed as channel information C
b, its Rch groups of filter coefficients is expressed as Rch groups of filter coefficients h
b (R), and its Lch groups of filter coefficients is expressed as Lch groups of filter coefficients h
b (L).
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 15 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 15, when loud speaker S is connected to output unit 4, coefficient setup unit 6 shows above-mentioned menu screen (St301) over the display.When receiving the operation input made by user, coefficient setup unit 6 selects the coefficient files (St302) of corresponding loud speaker.Subsequently, coefficient setup unit 6 is with reference to the channel information C (St303) be included in coefficient files.The R channel of this loud speaker and L channel have in the situation of different filter coefficient wherein, and coefficient setup unit 6 sets Rch groups of filter coefficients h in signal processing unit 3
(R)with Lch groups of filter coefficients h
(L)(St304).Or the R channel of loud speaker and L channel have in the situation of same filter coefficient wherein, coefficient setup unit 6 sets the groups of filter coefficients h (St304) shared by L channel and R channel in signal processing unit 3.When sending the instruction of reproducing audio, audio signal processor performs correction process with from loud speaker S output audio to audio signal as when the first embodiment in signal processing unit 3.
As mentioned above, in the present embodiment, coefficient files comprises channel information C, and the groups of filter coefficients h that this channel information C is used as about using in the L channel of corresponding loud speaker and R channel is identical or different information.Coefficient setup unit 6 is with reference to channel information C and set groups of filter coefficients h in digital filter.Thus, compared with the situation different with the loudspeaker performance wherein between R channel and L channel, groups of filter coefficients h can be reduced half in the situation that the L channel of loud speaker is identical with the loudspeaker performance of R channel wherein, and save the capacity of holding unit 5.
(the 4th embodiment)
Fourth embodiment of the present disclosure will be described now.
In the fourth embodiment, by the identical reference symbol sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the difference of the present embodiment and the first embodiment is the details of the coefficient files kept in holding unit 5.
[coefficient files]
Figure 16 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.As shown in figure 16, corresponding with each loud speaker coefficient files comprises groups of filter coefficients h and " number of channels " n.Here, loud speaker is that in the situation of stereo (two sound channels), coefficient files comprises the groups of filter coefficients h corresponding with corresponding sound channel wherein.In addition, loud speaker is that in the situation of monophony (sound channel), coefficient files comprises a groups of filter coefficients h wherein.Here, loud speaker S
bstereo, and loud speaker S
awith loud speaker S
cit is monophony.Number of channels n is stereo or monaural information about loud speaker.In figure 16, loud speaker S
anumber of channels be expressed as number of channels n
a, and its groups of filter coefficients is expressed as groups of filter coefficients h
a.This is equally applicable to loud speaker S
c.In addition, loud speaker S
bnumber of channels be expressed as number of channels n
b, its Rch groups of filter coefficients is expressed as Rch groups of filter coefficients h
b (R), and its Lch groups of filter coefficients is expressed as Lch groups of filter coefficients h
b (L).
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 17 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 17, when loud speaker S is connected to output unit 4, coefficient setup unit 6 shows above-mentioned menu screen (St401) over the display.When receiving the operation input made by user, coefficient setup unit 6 selects the coefficient files (St402) of corresponding loud speaker.Subsequently, coefficient setup unit 6 is with reference to the number of channels n be included in coefficient files (St403).The number of channels of loud speaker is in the situation of 2 wherein, that is, loud speaker is stereo, then coefficient setup unit 6 sets Rch groups of filter coefficients h in signal processing unit 3
(R)with Lch groups of filter coefficients h
(L)(St404).Or the number of channels of loud speaker is in the situation of 1 wherein, that is, loud speaker is monophony, then coefficient setup unit 6 sets Rch groups of filter coefficients h in signal processing unit 3
(R)with Lch groups of filter coefficients h
(L)in one (St404).When sending the instruction of reproducing audio, audio signal processor performs correction process with from loud speaker S output audio to audio signal as when the first embodiment in signal processing unit 3.
As mentioned above, in the present embodiment, coefficient files comprises number of channels n, and this number of channels n is used as the information of the number of channels of corresponding loud speaker.Coefficient setup unit 6 is with reference to number of channels n and set groups of filter coefficients h in digital filter.Loud speaker is in monaural situation wherein, can adjust number of channels for digital filter process to reduce amount of calculation.In addition, compared with being stereosonic situation with wherein loud speaker, loud speaker groups of filter coefficients h can be reduced half in monaural situation wherein, and save the capacity of holding unit 5.
(the 5th embodiment)
Fifth embodiment of the present disclosure will be described now.
In the 5th embodiment, by the identical reference symbol sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the difference of the present embodiment and the first embodiment is the details of the coefficient files kept in holding unit 5.In addition, in the present embodiment, the type information of the information of model, model number etc. is indicated to be endowed loud speaker S.
[coefficient files]
Figure 18 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.As shown in figure 18, corresponding with each loud speaker coefficient files comprises " loud speaker identification information " i.Loud speaker identification information i be for compared with the loud speaker type information obtained from connected loud speaker S to search for the information of coefficient of correspondence file.In figure 18, loud speaker S
aloud speaker identification information be expressed as loud speaker identification information i
a, loud speaker S
bloud speaker identification information be expressed as loud speaker identification information i
b, loud speaker S
cloud speaker identification information be expressed as loud speaker identification information i
c.
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 19 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 19, when loud speaker S is connected to output unit 4, coefficient setup unit 6 obtains the type information (St501) of loud speaker S.Next, the type information of loud speaker S compared with the loud speaker identification information i be included in each coefficient files, and is specified the coefficient files (St502) corresponding with loud speaker S by coefficient setup unit 6.Subsequently, coefficient setup unit 6 sets the groups of filter coefficients h (St503) be included in coefficient files in the digital filter F of signal processing unit 3.When sending the instruction of reproducing audio, audio signal processor performs correction process with from loud speaker S output audio to audio signal as when the first embodiment in signal processing unit 3.
As mentioned above, in the present embodiment, coefficient files comprises loud speaker identification information i, and this loud speaker identification information i is used as the search coefficient files corresponding with loud speaker S.Correspondingly, when loud speaker S connects, the audio signal processor according to the present embodiment can set the groups of filter coefficients h corresponding with loud speaker S automatically, and without the need to receiving the operation input made by user.
(the 6th embodiment)
Sixth embodiment of the present disclosure will be described now.
In the sixth embodiment, by the identical reference symbol sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the difference of the present embodiment and the first embodiment is the details of the coefficient files kept in holding unit 5.
[coefficient files]
Figure 20 is the schematic diagram of the coefficient files that the various loud speakers kept in holding unit 5 are shown.As shown in figure 20, corresponding with each loud speaker coefficient files comprises " coefficient word length " p.Coefficient word length p is used for describing the word length for the coefficient of the signal transacting in signal processing unit 3, such as 16 or 32.In fig. 20, loud speaker S
acoefficient word length represent loud speaker identification information p
a, loud speaker S
bcoefficient word length represent loud speaker identification information p
b, loud speaker S
ccoefficient word length represent loud speaker identification information p
c.
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 21 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 21, when loud speaker S is connected to output unit 4, coefficient setup unit 6 shows above-mentioned menu screen (St601) over the display.When receiving the operation input made by user, coefficient setup unit 6 selects the coefficient files (St602) of corresponding loud speaker.Subsequently, coefficient setup unit 6 is with reference to the coefficient word length p (St603) be included in coefficient files.In addition, coefficient setup unit 6 sets the groups of filter coefficients h (St604) be included in selected coefficient files in signal processing unit 3.When sending the instruction of reproducing audio, audio signal processor coefficient of utilization word length p performs correction process with from loud speaker S output audio to audio signal in signal processing unit 3.
As mentioned above, in the present embodiment, coefficient files comprises coefficient word length p, and this coefficient word length p is used as the word length of the coefficient of the signal transacting in information process unit 3.Correspondingly, the amount of calculation in signal processing unit 3 can be reduced.
(the 7th embodiment)
Seventh embodiment of the present disclosure will be described now.
In the 7th embodiment, by the identical reference symbol sign structure identical with the first embodiment and by the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the audio signal processor according to the present embodiment and the difference of the audio signal processor 1 according to the first embodiment are that audio signal processor itself can create the groups of filter coefficients of be connected loud speaker wherein.
[structure of audio signal processor]
Figure 22 is the block diagram of the audio signal processor 20 illustrated according to embodiment of the present disclosure.As shown in figure 22, except the structure of the audio signal processor 1 according to the first embodiment, audio signal processor 20 also comprises coefficient generation unit 21 and microphone 22.Microphone 22 is connected to coefficient generation unit 21 and coefficient generation unit 21 is connected to holding unit 5.
The audio frequency exported from loud speaker S collected by microphone 22, so that audio frequency is sent to coefficient generation unit 21.Groups of filter coefficients h according to the groups of filter coefficients h of the audio computer loud speaker S collected by microphone 22, and is stored in coefficient files to hold it in holding unit 5 by coefficient generation unit 21.Coefficient generation unit 21 comprises the A/D converter audio signal of being collected by microphone 22 being performed to A/D conversion.
Figure 23 is the perspective view of the outward appearance that audio signal processor 20 is shown.As shown in figure 23, audio signal processor 20 is connected to loud speaker S.Figure 24 illustrates the following state of audio signal processor 20, is wherein collected the audio frequency exported from loud speaker S by microphone 22.In addition, as shown in figure 25, microphone 22 can be dismountable from audio signal processor 20.
[interpolation of coefficient files]
When the loud speaker S that its coefficient files does not remain in holding unit 5 is connected to audio signal processor 20, audio signal processor 20 exports test signal from output unit 4 to loud speaker S.This test signal can be above-mentioned impulse signal.Microphone 22 is collected by test signal from the audio frequency that loud speaker S exports, and this audio frequency is sent to coefficient generation unit 21.
Coefficient generation unit 21 is according to the audio frequency collected by microphone 22 (impulse response) calculating filter coefficient group h.Groups of filter coefficients h can be calculated by said method.Calculated groups of filter coefficients h is supplied to holding unit 5 by coefficient generation unit 21.In this case, groups of filter coefficients h is stored in the coefficient files be associated with the model of loud speaker S to be remained in holding unit 5 by groups of filter coefficients h by coefficient generation unit 21.The model of loud speaker S can be inputted or can be used in the loud speaker identification information i described in the 5th embodiment and obtain by user.In this way, be connected in the situation of audio signal processor 20 at its coefficient files loud speaker do not remained in holding unit 5, audio signal processor 20 itself can add the coefficient files of this loud speaker.
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 26 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 26, when loud speaker S is connected to output unit 4, coefficient setup unit 6 searches for holding unit 5 to check whether the coefficient files (St701) of the loud speaker model keeping corresponding with loud speaker S.If maintain the coefficient files (St702: yes) of loud speaker S in holding unit 5, then coefficient setup unit 6 selects this coefficient files (St703).If do not keep the coefficient files of loud speaker S (St702: no) in holding unit 5, then coefficient setup unit 6 measures the impulse response (St704) of loud speaker S.Coefficient generation unit 21 calculates the groups of filter coefficients h (St705) of loud speaker S based on measured impulse response, and adds the coefficient files comprising groups of filter coefficients h to holding unit 5 (St706).Then coefficient setup unit 6 selects the coefficient files (St703) that adds.
Coefficient setup unit 6 sets the groups of filter coefficients h (St707) be included in the coefficient files selected in St703 in signal processing unit 3.When sending the instruction of reproducing audio, audio signal processor uses the groups of filter coefficients h that is included in coefficient files to perform correction process to audio signal in signal processing unit 3, with from loud speaker S output audio.
As mentioned above, in the present embodiment, even if when its coefficient files loud speaker do not remained in holding unit 5 is connected to audio signal processor 20, audio signal processor 20 also can add the coefficient files of this loud speaker to holding unit 5.Correspondingly, even if when its coefficient files loud speaker do not remained in holding unit 5 is connected to audio signal processor 20, audio signal processor 20 also can correct the loudspeaker performance of this loud speaker.
(the 8th embodiment)
Eighth embodiment of the present disclosure will be described now.
In the 8th embodiment, indicate the structure identical with the first embodiment and the 7th embodiment by identical reference symbol and general the descriptions thereof are omitted.
Something in common according to the audio signal processor of the present embodiment and the audio signal processor of the first embodiment is, coefficient setup unit 6 selects the groups of filter coefficients h corresponding with the model of the loud speaker that will be connected to output unit 4 from holding unit 5, and in signal processing unit 3, groups of filter coefficients h is used for correction process.But the audio signal processor according to the present embodiment and the difference of the audio signal processor 1 according to the first embodiment are that be connected loud speaker is associated with the similarity factor file kept in holding unit 5 by audio signal processor.
[association of coefficient files]
When the loud speaker S that its coefficient files does not remain in holding unit 5 is connected to audio signal processor 20, audio signal processor 20 exports test signal from output unit 4 to loud speaker S.This test signal can be above-mentioned impulse signal.Microphone 22 is collected by test signal from the audio frequency that loud speaker S exports, and this audio frequency is sent to coefficient generation unit 21.
Coefficient generation unit 21 is according to the audio frequency collected by microphone 22 (impulse response) calculating filter coefficient group h.Groups of filter coefficients h can be calculated by said method.Next, compared with the groups of filter coefficients h that the groups of filter coefficients h calculated and the coefficient files of the various loud speakers kept in holding unit 5 comprise by coefficient generation unit 21.Then, new loud speaker is associated with the coefficient files comprising the groups of filter coefficients h with maximum similarity by coefficient generation unit 21 further.Here, " be associated " the new loud speaker referring to and change the coefficient files corresponding with existing loud speaker and make to support to add.
[operation of audio signal processor]
The operation according to the audio signal processor of the present embodiment will be described now.
Figure 27 is the flow chart of the operation that audio signal processor is shown.
As shown in figure 27, when loud speaker S is connected to output unit 4, coefficient setup unit 6 searches for holding unit 5 to check whether the coefficient files (St801) of the loud speaker model keeping corresponding with loud speaker S.If maintain the coefficient files (St802: yes) of loud speaker S in holding unit 5, then coefficient setup unit 6 selects this coefficient files (St803).If do not keep the coefficient files of loud speaker S (St802: no) in holding unit 5, then coefficient setup unit 6 measures the impulse response (St804) of loud speaker S.Coefficient generation unit 21 calculates the groups of filter coefficients h (St805) of loud speaker S based on measured impulse response.Next, the groups of filter coefficients that the coefficient files of the various loud speakers kept in the groups of filter coefficients h calculated and holding unit 5 comprises compares by coefficient generation unit 21, and new loud speaker is associated (St806) with the coefficient files comprising the groups of filter coefficients h with maximum similarity.Coefficient setup unit 6 selects the coefficient files (St803) added.
Coefficient setup unit 6 sets the groups of filter coefficients h (St807) be included in the coefficient files selected in St803 in signal processing unit 3.When sending the instruction of reproducing audio, audio signal processor uses the groups of filter coefficients h that is included in coefficient files to perform correction process to audio signal in signal processing unit 3, with from loud speaker S output audio.
As mentioned above, in the present embodiment, even if when its coefficient files loud speaker do not remained in holding unit 5 is connected to audio signal processor 20, the coefficient files of loud speaker also can be associated with the coefficient files kept in holding unit 5 by audio signal processor 20.Correspondingly, even if when its coefficient files loud speaker do not remained in holding unit 5 is connected to audio signal processor 20, audio signal processor 20 also can correct the loudspeaker performance of this loud speaker.Here, owing to using existing coefficient files as the coefficient files of new loud speaker and the coefficient files of new loud speaker does not remain in holding unit 5, so can save the capacity of holding unit 5.
The disclosure is not limited to above-described embodiment, but can carry out various change when not departing from spirit of the present disclosure.
In the above-described embodiments, signal processing unit 3 corrects the loudspeaker performance of loud speaker.In addition, signal processing unit 3 can perform the correction process of the audio effect processing adding such as virtual sound image location and so on to audio signal.
Disclosed in the Japanese Priority Patent application JP2010-126798 that the theme that the disclosure comprises and on June 2nd, 2010 submit to Japan Office, theme is relevant, and its full content is by reference to being incorporated in this.
It will be understood by those skilled in the art that and can carry out various amendment, combination, sub-portfolio and replacement according to designing requirement and other factors, as long as these amendments, combination, sub-portfolio and replacement are in the scope of claims and equivalent thereof.
According to foregoing, can find out, The embodiment provides following technical scheme.
A kind of audio signal processor, comprising: signal processing unit, is configured to pass digital filter to the process of audio signal executive signal; Output unit, is configured to be connected to outside loud speaker and described audio signal is outputted to described loud speaker; Holding unit, is configured to keep multiple filter coefficient, and described multiple filter coefficient is the impulse response of the opposite characteristic of multiple loud speakers with different loudspeaker performance; And coefficient setup unit, be configured to select the filter coefficient corresponding with the described loud speaker being connected to described output unit from described holding unit and set filter coefficient described digital filter.
A kind of acoustic signal processing method, comprising: measure the impulse response with multiple loud speakers of different loudspeaker performance; The filter coefficient obtained from described impulse response is remained in holding unit, described filter coefficient is associated with described multiple loud speaker simultaneously; And from described holding unit, select filter coefficient corresponding to loud speaker with connection, to set filter coefficient and by described filter coefficient applied audio signal in digital filter.
Claims (9)
1. an audio signal processor, comprising:
Signal processing unit, is configured to pass digital filter to the process of audio signal executive signal;
Output unit, is configured to be connected to outside loud speaker and the loud speaker described audio signal being outputted to described outside;
Holding unit, is configured to keep multiple filter coefficient, and described multiple filter coefficient is the impulse response of the opposite characteristic of multiple loud speakers with different loudspeaker performance; And described holding unit is configured to keep the coefficient word length corresponding with each loud speaker in described multiple loud speaker; And
Coefficient setup unit, be configured to from described multiple filter coefficient, select the filter coefficient corresponding with the loud speaker of the described outside being connected to described output unit based on described coefficient word length, and be configured to set described filter coefficient in described digital filter.
2. audio signal processor according to claim 1, wherein
Described holding unit is configured to keep the coefficient length of each in described multiple filter coefficient corresponding with the reproduction band of described multiple loud speaker, and
Described coefficient setup unit is configured in described digital filter, set filter coefficient with reference to described coefficient length.
3. audio signal processor according to claim 1, wherein
Described coefficient setup unit is configured in described digital filter, set filter coefficient with reference to sound channel set information.
4. audio signal processor according to claim 1, wherein
Described holding unit is configured to keep number of channels information, and described number of channels information is corresponding with each loud speaker of described multiple loud speaker and indicate number of channels, and
Described coefficient setup unit is configured in described digital filter, set filter coefficient with reference to described number of channels information.
5. audio signal processor according to claim 1, wherein
Described holding unit is configured to keep loud speaker identification information, and described loud speaker identification information is corresponding with each loud speaker in described multiple loud speaker and be associated with each model of described multiple loud speaker, and
Described coefficient setup unit is configured to setting in described digital filter and is assigned with the filter coefficient of the loud speaker of the outside of the loud speaker identification information corresponding to out of Memory, and the described out of Memory wherein obtained from the loud speaker of the outside being connected to described output unit indicates the model of the loud speaker of described outside.
6. audio signal processor according to claim 1, wherein
Described coefficient setup unit is configured in described digital filter, set filter coefficient with reference to described coefficient word length.
7. audio signal processor according to claim 1, also comprises:
Test signal output unit, the loud speaker be configured to the outside being connected to described output unit exports test signal;
Audio collection unit, is configured to pass described test signal and collects the audio frequency exported from the loud speaker of described outside; And
Coefficient generation unit, the audio frequency being configured to collect from described audio collection unit exports and generates the filter coefficient corresponding with the loud speaker of described outside, and keeps described filter coefficient in described holding unit.
8. audio signal processor according to claim 1, also comprises:
Test signal output unit, the loud speaker be configured to the outside being connected to described output unit exports test signal;
Audio collection unit, is configured to pass described test signal and collects the audio frequency exported from the loud speaker of described outside; And
Coefficient generation unit, the audio frequency being configured to collect from described audio collection unit exports and generates the filter coefficient corresponding with the loud speaker of described outside, and is associated with the filter coefficient with highest similarity in the described multiple filter coefficient kept in described holding unit by the loud speaker of described outside.
9. an acoustic signal processing method, comprising:
Measure the impulse response with multiple loud speakers of different loudspeaker performance;
Keep the multiple filter coefficients obtained from described impulse response, described multiple filter coefficient is associated with described multiple loud speaker simultaneously;
Keep the coefficient word length corresponding with each loud speaker in described multiple loud speaker; And
Select from described multiple filter coefficient and filter coefficient corresponding to loud speaker connected based on described coefficient word length, to set filter coefficient and by described filter coefficient applied audio signal in digital filter.
Applications Claiming Priority (2)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP2010126798A JP5598095B2 (en) | 2010-06-02 | 2010-06-02 | Audio signal processing apparatus and audio signal processing method |
| JP2010-126798 | 2010-06-02 |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN102316406A CN102316406A (en) | 2012-01-11 |
| CN102316406B true CN102316406B (en) | 2015-06-17 |
Family
ID=45065079
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201110151059.7A Active CN102316406B (en) | 2010-06-02 | 2011-05-26 | Audio signal processing apparatus and audio signal processing method |
Country Status (3)
| Country | Link |
|---|---|
| US (1) | US9264800B2 (en) |
| JP (1) | JP5598095B2 (en) |
| CN (1) | CN102316406B (en) |
Families Citing this family (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US9130527B2 (en) * | 2010-08-18 | 2015-09-08 | Dolby Laboratories Licensing Corporation | Method and system for controlling distortion in a critical frequency band of an audio signal |
| US9020623B2 (en) | 2012-06-19 | 2015-04-28 | Sonos, Inc | Methods and apparatus to provide an infrared signal |
| JP5598589B1 (en) * | 2013-10-16 | 2014-10-01 | オンキヨー株式会社 | Equalizer device and equalizer program |
| WO2016103810A1 (en) * | 2014-12-26 | 2016-06-30 | パイオニア株式会社 | Sound reproducing device |
| US9678707B2 (en) | 2015-04-10 | 2017-06-13 | Sonos, Inc. | Identification of audio content facilitated by playback device |
| EP3890347A1 (en) * | 2015-09-10 | 2021-10-06 | Yayuma Audio SP. Z.O.O. | A method of an audio signal correction |
| US10356526B2 (en) | 2015-09-28 | 2019-07-16 | Razer (Asia-Pacific) Pte. Ltd. | Computers, methods for controlling a computer, and computer-readable media |
| CN105554622A (en) * | 2016-03-16 | 2016-05-04 | 朱丽芬 | Low-noise loudspeaker |
| US10856303B2 (en) * | 2017-10-06 | 2020-12-01 | Lg Electronics Inc. | Method for performing measurement and device supporting the same |
| US11570543B2 (en) * | 2021-01-21 | 2023-01-31 | Biamp Systems, LLC | Loudspeaker polar pattern creation procedure |
Citations (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6674864B1 (en) * | 1997-12-23 | 2004-01-06 | Ati Technologies | Adaptive speaker compensation system for a multimedia computer system |
Family Cites Families (9)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JPH0888893A (en) * | 1994-09-20 | 1996-04-02 | Victor Co Of Japan Ltd | Audio signal transmission circuit |
| JPH08110783A (en) * | 1994-10-11 | 1996-04-30 | Victor Co Of Japan Ltd | Audio signal transmission circuit and convolver coefficient arithmetic unit |
| US7463170B2 (en) * | 2006-11-30 | 2008-12-09 | Broadcom Corporation | Method and system for processing multi-rate audio from a plurality of audio processing sources |
| JP2008228133A (en) * | 2007-03-15 | 2008-09-25 | Matsushita Electric Ind Co Ltd | Acoustic system |
| US8077880B2 (en) * | 2007-05-11 | 2011-12-13 | Audyssey Laboratories, Inc. | Combined multirate-based and fir-based filtering technique for room acoustic equalization |
| JP5018339B2 (en) | 2007-08-23 | 2012-09-05 | ソニー株式会社 | Signal processing apparatus, signal processing method, and program |
| WO2009078176A1 (en) * | 2007-12-19 | 2009-06-25 | Panasonic Corporation | Video/audio output system |
| US8401202B2 (en) * | 2008-03-07 | 2013-03-19 | Ksc Industries Incorporated | Speakers with a digital signal processor |
| JP2011064961A (en) * | 2009-09-17 | 2011-03-31 | Toshiba Corp | Audio playback device and method |
-
2010
- 2010-06-02 JP JP2010126798A patent/JP5598095B2/en not_active Expired - Fee Related
-
2011
- 2011-05-17 US US13/109,166 patent/US9264800B2/en not_active Expired - Fee Related
- 2011-05-26 CN CN201110151059.7A patent/CN102316406B/en active Active
Patent Citations (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6674864B1 (en) * | 1997-12-23 | 2004-01-06 | Ati Technologies | Adaptive speaker compensation system for a multimedia computer system |
Also Published As
| Publication number | Publication date |
|---|---|
| JP5598095B2 (en) | 2014-10-01 |
| US20110301731A1 (en) | 2011-12-08 |
| JP2011254297A (en) | 2011-12-15 |
| US9264800B2 (en) | 2016-02-16 |
| CN102316406A (en) | 2012-01-11 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN102316406B (en) | Audio signal processing apparatus and audio signal processing method | |
| JP4466658B2 (en) | Signal processing apparatus, signal processing method, and program | |
| CN100541145C (en) | Sound measuring apparatus and method and audio signal processing apparatus | |
| EP3026931B1 (en) | Method, system and appraratus for loudspeaker excursion domain processing | |
| US9414161B2 (en) | Method, system and apparatus for loudspeaker excursion domain processing | |
| CN103634726B (en) | A kind of Automatic loudspeaker equalization method | |
| EP2959579B1 (en) | Equalization filter coefficient determinator, apparatus, equalization filter coefficient processor, system and methods | |
| CN106535039A (en) | Loudness-Based Audio-Signal Compensation | |
| US7286946B2 (en) | Transmission characteristic measuring device transmission characteristic measuring method, and amplifier | |
| CN104604254A (en) | Audio processing device, method, and program | |
| CN103517199A (en) | Apparatus and method for localizing sound image | |
| CN103702253A (en) | Information processing method and electronic equipment | |
| JP2020528245A (en) | Audio playback method and multi-screen terminal | |
| JP5682539B2 (en) | Sound playback device | |
| CN105188008A (en) | Method and device for testing audio output unit | |
| CN113595527A (en) | Filtering parameter determination method, filtering method and related device | |
| US8401198B2 (en) | Method of improving acoustic properties in music reproduction apparatus and recording medium and music reproduction apparatus suitable for the method | |
| CN112449286B (en) | System and method for complementary audio output | |
| US9992573B1 (en) | Phase inversion filter for correcting low frequency phase distortion in a loudspeaker system | |
| CN108540905B (en) | Frequency divider circuit and frequency divider | |
| CN104640050A (en) | Horn resonant frequency measuring method and measuring device thereof | |
| CN105338462A (en) | Implementation method for reproducing hearing-aid insertion gain | |
| CN109545174A (en) | A kind of audio-frequency processing method, device and equipment | |
| US10362396B2 (en) | Phase control signal generation device, phase control signal generation method, and phase control signal generation program | |
| US20110317853A1 (en) | Heyser spiral low frequency correction of fir filters |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant |